1. Field of the Invention
The invention relates to audio processing systems and particularly customized audio adjustment systems.
2. Description of the Related Technology
Personal audio players are nearly ubiquitous. The popularization of smartphones has ushered in an environment where anyone and everyone with a smartphone has an on-board personal audio player. Personal audio is typically delivered to a user by headphones. Headphones are a pair of small speakers that are designed to be held in place close to a user's ears. They may be electroacoustic transducers which convert an electrical signal to a corresponding sound in the user's ear. Headphones are designed to allow a single user to listen to an audio source privately, in contrast to a loudspeaker which emits sound into the open air, allowing anyone nearby to listen. Earbuds or earphones are in-ear versions of headphones.
Active noise reduction; active noise cancellation and active noise control are known in the prior art Elliot, S. J. et al., “Active Noise Control,” IEEE Signal Processing Magazine, October 1993 (pages 12-35), the disclosure of which is expressly incorporated by reference herein, describes the history and background of active noise control systems and describes the use of adaptive filters.
Kuo, Sen M. et al., “Active Noise Control: A Tutorial Review,” Proceeding of the IEEE, Vol. 87, No. 6, June 1999 (pages 943-973), the disclosure of which is expressly incorporated by reference herein, describes principles and systems for active noise control.
Kuo, Sen M. et al., “Design of Active Noise Control Systems with the TMS320 Family,” Application Report, Texas Instruments Digital Signal Processing Solutions, Digital Signal Processing Products-Semiconductor Group, SPRA042, June 1996, the disclosure of which is expressly incorporated by reference herein, describes specialized digital signal processors designed for real-time processing of digitized signals and details the design of an Active Noise Control (“ANC”) system using a TMS320 DSP.
United States Published Patent Application US 2014-0044275, the disclosure of which is expressly incorporated by reference herein, describes an active noise control system with compensation for error sensing at the ear drum including a subjective tuning module and user control.
Active noise control systems utilize various active filtration techniques and rely on algorithms to process source audio in order to reduce the influence of noise on the listener. This may be accompanied by modification of the source audio by combination with an “anti-noise” signal derived from comparing ambient sound to source audio at the ear of a listener.
Active noise control devices in the prior art suffer from being incapable of addressing the wide variation of ambient sound, dominant noise, acoustic sensors, specific characteristics of headphones or earphones or other listening devices, the type nature and characteristics of source audio (such as sound from a digital electronic device), and individual audio perceptions as each of these and other elements of sound interact to comprise a listening experience.
Adaptive noise cancellation is described in Singh, Arti. “Adaptive Noise Cancellation,” Dept. of Electronics & Communications, Netaji Subhas Institute of Technology, (2001). http://www.cs.cmu.edu/naarti/pubs/ANC.pdf#. Accessed Nov. 21, 2014, the disclosure of which is incorporated herein. The customization according to the invention may be performed in accordance with the principles described therein.
U.S. Patent Application Publication No. US 2013/0325993 A1, the disclosure of which is incorporated by reference herein, discloses a method and system for group-based communication in a social networking space. The system is for managing and tracking social networking group events and does not contemplate free form connections for audio communications.
Advancements in hearing aid technology have resulted in numerous developments which have served to improve the listening experience for people with hearing impairments, but these developments have been fundamentally limited by an overriding need to minimize size and maximize invisibility of the device. Resulting limitations from miniaturized form factors include limits on battery size and life, power consumption and, thus, processing power, typically two or fewer microphones per side (left and right) and a singular focus on speech recognition and speech enhancement.
Hearing aid technology may use “beamforming” and other methods to allow for directional sound targeting to isolate and amplify just speech, wherever that speech might be located.
Hearing aid technology includes methods and apparatus to isolate and amplify speech and only speech, in a wide variety of environments, focusing on the challenge of “speech in noise” or the “cocktail party” effect (the use of directional sound targeting in combination with noise cancellation has been the primary approach to this problem).
Hearing aid applications typically ignore or minimize any sound in the ambient environment other than speech. Hearing devices may also feature artificial creation of sounds as masking to compensate for tinnitus or other unpleasant remnants of the assistive listening experience for those suffering from hearing loss.
Due to miniature form factors, hearing aids are constrained by a severe restriction on available power to preserve battery life which results in limitations in signal processing power. Applications and devices not constrained by such limitations but rather focused on providing the highest quality listening experience are able to utilize the highest quality of signal processing, which among other things, will maintain a high sampling rate, typically at least twice that of the highest frequency that can be perceived. Music CDs have a 44.1 kHz sampling rate to preserve the ability to process sound with frequencies up to about 20 kHz. Most hearing devices sample at rates significantly below 44.1 kHz, resulting in a much lower range of frequencies that can be analyzed for speech patterns and then amplified, further necessitating the use of compression and other compensating methodologies in an effort to preserve the critical elements of speech recognition and speech triggers that reside in higher frequencies.
Hearing aids have almost always required the need to compensate for loss of hearing at very high frequencies, and given equivalent volume is much higher for very high and very low frequencies (i.e., more amplification is required to achieve a similar volume in higher and lower frequencies as midrange frequencies), one strategy has been compression (wide dynamic range compression or WDRC) whereby either the higher frequency ranges are compressed to fit within a lower frequency band, or less beneficially, higher frequency ranges are literally cut and pasted into a lower band, which requires a learning curve for the user.
For these reasons hearing aid technologies do not adequately function within the higher frequency bands where a great deal of desired ambient sound exists for listeners, and hearing aids and their associated technologies have neither been developed to, nor are capable as developed, to enhance the listening experience for listeners who do not suffer from hearing loss but rather want an optimized listening experience.
Noise reduction systems have been implemented in such a way that their use and processing is fixed across listening environments in either an On/Off paradigm or a degree of noise reduction setting, or on a frequency-specific basis utilizing multi-channel processors to apply noise reduction within specific frequency bands, however, in each of these systems, other than identifying speech within a hearing aid application, these noise reduction systems have treated all ambient noise as a single class of disturbance.
Typical hearing devices utilize either a system of a) isolating steady-state sound or other ambient sounds that do not correspond to predetermined modulation rates and peak to trough characteristics or b) measure signal to noise ratios in an ambient environment which all assume the desired “signal” is speech, or within frequency bands in a multi-channel system to similarly isolate environments in which signal to noise ratios are high (all ambient sound is not too loud and thus lower or no noise suppression across frequencies or within frequency bands is applied) or in which signal to noise ratios are low (all ambient sound is deemed to be too loud/undesirable and thus more noise suppression is applied), but the invention will allow similar systems to be employed with the fundamental and unique attribute that they will allow the listener to determine which sounds or signals in the ambient environment are desirable and to similarly determine which signals or sound profiles constitute undesired noise, thus enabling the established methodologies of utilizing modulation and other sound pattern or signal to noise methodologies to be employed in the current invention. These methodologies may incorporate the inclusion of speech, in general, as the relevant signal, or may further refine the characteristics of that speech to associate the signal with the speech of a child or of children, or certain specific individuals or sounds which incorporate speech as part of their acoustic signal, but will also focus on the limitless combination of ambient sound which are, in fact, desirable and not group all such sounds into a single group as has been done in the prior art. Headphone, earphone and other listening devices have focused on the reproduction of source audio signals at the ears of listeners and have all been developed with the assumption or belief that such source audio signal is the only source of desired sound. These listening devices later incorporated one or more microphones either for use in noise cancellation or to enable the listening devices to function as the speaking and hearing components of wireless communication devices, recognizing the benefit to users of not having to remove such listening device when using such wireless communication system. In each of these incarnations and scenarios where users may wish to communicate with others in their presence, these listening devices have muted the source sound while activating the microphone. Neither hearing aid nor active noise cancellation technologies are capable of permitting users to communicate with others in their presence while also permitting admission of desirable audio information to the user.
It is known to use microphone arrays and beamforming technology in order to locate and isolate an audio source. Personal audio is typically delivered to a user by headphones. Headphones are a pair of small speakers that are designed to be held in place close to a user's ears. They may be electroacoustic transducers which convert an electrical signal to a corresponding sound in the user's ear. Headphones are designed to allow a single user to listen to an audio source privately, in contrast to a loudspeaker which emits sound into the open air, allowing anyone nearby to listen. Earbuds or earphones are in-ear versions of headphones.
A sensitive transducer element of a microphone is called its element or capsule. Except in thermophone based microphones, sound is first converted to mechanical motion by means of a diaphragm, the motion of which is then converted to an electrical signal. A complete microphone also includes a housing, some means of bringing the signal from the element to other equipment, and often an electronic circuit to adapt the output of the capsule to the equipment being driven. A wireless microphone contains a radio transmitter.
The condenser microphone, is also called a capacitor microphone or electrostatic microphone. Here, the diaphragm acts as one plate of a capacitor, and the vibrations produce changes in the distance between the plates.
A fiber optic microphone converts acoustic waves into electrical signals by sensing changes in light intensity, instead of sensing changes in capacitance or magnetic fields as with conventional microphones. During operation, light from a laser source travels through an optical fiber to illuminate the surface of a reflective diaphragm. Sound vibrations of the diaphragm modulate the intensity of light reflecting off the diaphragm in a specific direction. The modulated light is then transmitted over a second optical fiber to a photo detector, which transforms the intensity-modulated light into analog or digital audio for transmission or recording. Fiber optic microphones possess high dynamic and frequency range, similar to the best high fidelity conventional microphones. Fiber optic microphones do not react to or influence any electrical, magnetic, electrostatic or radioactive fields (this is called EMI/RFI immunity). The fiber optic microphone design is therefore ideal for use in areas where conventional microphones are ineffective or dangerous, such as inside industrial turbines or in magnetic resonance imaging (MRI) equipment environments.
Fiber optic microphones are robust, resistant to environmental changes in heat and moisture, and can be produced for any directionality or impedance matching. The distance between the microphone's light source and its photo detector may be up to several kilometers without need for any preamplifier or other electrical device, making fiber optic microphones suitable for industrial and surveillance acoustic monitoring. Fiber optic microphones are suitable for use application areas such as for infrasound monitoring and noise-canceling.
U.S. Pat. No. 6,462,808 B2, the disclosure of which is incorporated by reference herein shows a small optical microphone/sensor for measuring distances to, and/or physical properties of, a reflective surface
The MEMS (MicroElectrical-Mechanical System) microphone is also called a microphone chip or silicon microphone. A pressure-sensitive diaphragm is etched directly into a silicon wafer by MEMS processing techniques, and is usually accompanied with integrated preamplifier. Most MEMS microphones are variants of the condenser microphone design. Digital MEMS microphones have built in analog-to-digital converter (ADC) circuits on the same CMOS chip making the chip a digital microphone and so more readily integrated with modern digital products. Major manufacturers producing MEMS silicon microphones are Wolfson Microelectronics (WM7xxx), Analog Devices, Akustica (AKU200x), Infineon (SMM310 product), Knowles Electronics, Memstech (MSMx), NXP Semiconductors, Sonion MEMS, Vesper, AAC Acoustic Technologies, and Omron.
A microphone's directionality or polar pattern indicates how sensitive it is to sounds arriving at different angles about its central axis. The polar pattern represents the locus of points that produce the same signal level output in the microphone if a given sound pressure level (SPL) is generated from that point. How the physical body of the microphone is oriented relative to the diagrams depends on the microphone design. Large-membrane microphones are often known as “side fire” or “side address” on the basis of the sideward orientation of their directionality. Small diaphragm microphones are commonly known as “end fire” or “top/end address” on the basis of the orientation of their directionality.
Some microphone designs combine several principles in creating the desired polar pattern. This ranges from shielding (meaning diffraction/dissipation/absorption) by the housing itself to electronically combining dual membranes.
An omni-directional (or non-directional) microphone's response is generally considered to be a perfect sphere in three dimensions. In the real world, this is not the case. As with directional microphones, the polar pattern for an “omni-directional” microphone is a function of frequency. The body of the microphone is not infinitely small and, as a consequence, it tends to get in its own way with respect to sounds arriving from the rear, causing a slight flattening of the polar response. This flattening increases as the diameter of the microphone (assuming it's cylindrical) reaches the wavelength of the frequency in question.
A unidirectional microphone is sensitive to sounds from only one direction
A noise-canceling microphone is a highly directional design intended for noisy environments. One such use is in aircraft cockpits where they are normally installed as boom microphones on headsets. Another use is in live event support on loud concert stages for vocalists involved with live performances. Many noise-canceling microphones combine signals received from two diaphragms that are in opposite electrical polarity or are processed electronically. In dual diaphragm designs, the main diaphragm is mounted closest to the intended source and the second is positioned farther away from the source so that it can pick up environmental sounds to be subtracted from the main diaphragm's signal. After the two signals have been combined, sounds other than the intended source are greatly reduced, substantially increasing intelligibility. Other noise-canceling designs use one diaphragm that is affected by ports open to the sides and rear of the microphone.
Sensitivity indicates how well the microphone converts acoustic pressure to output voltage. A high sensitivity microphone creates more voltage and so needs less amplification at the mixer or recording device. This is a practical concern but is not directly an indication of the microphone's quality, and in fact the term sensitivity is something of a misnomer, “transduction gain” being perhaps more meaningful, (or just “output level”) because true sensitivity is generally set by the noise floor, and too much “sensitivity” in terms of output level compromises the clipping level.
A microphone array is any number of microphones operating in tandem. Microphone arrays may be used in systems for extracting voice input from ambient noise (notably telephones, speech recognition systems, hearing aids), surround sound and related technologies, binaural recording, locating objects by sound: acoustic source localization, e.g., military use to locate the source(s) of artillery fire, aircraft location and tracking.
Typically, an array is made up of omni-directional microphones, directional microphones, or a mix of omni-directional and directional microphones distributed about the perimeter of a space, linked to a computer that records and interprets the results into a coherent form. Arrays may also be formed using numbers of very closely spaced microphones. Given a fixed physical relationship in space between the different individual microphone transducer array elements, simultaneous DSP (digital signal processor) processing of the signals from each of the individual microphone array elements can create one or more “virtual” microphones.
Beamforming or spatial filtering is a signal processing technique used in sensor arrays for directional signal transmission or reception. This is achieved by combining elements in a phased array in such a way that signals at particular angles experience constructive interference while others experience destructive interference. A phased array is an array of antennas, microphones, or other sensors in which the relative phases of respective signals are set in such a way that the effective radiation pattern is reinforced in a desired direction and suppressed in undesired directions. The phase relationship may be adjusted for beam steering. Beamforming can be used at both the transmitting and receiving ends in order to achieve spatial selectivity. The improvement compared with omni-directional reception/transmission is known as the receive/transmit gain (or loss).
Adaptive beamforming is used to detect and estimate a signal-of-interest at the output of a sensor array by means of optimal (e.g., least-squares) spatial filtering and interference rejection.
To change the directionality of the array when transmitting, a beamformer controls the phase and relative amplitude of the signal at each transmitter, in order to create a pattern of constructive and destructive interference in the wavefront. When receiving, information from different sensors is combined in a way where the expected pattern of radiation is preferentially observed.
With narrow-band systems the time delay is equivalent to a “phase shift”, so in the case of a sensor array, each sensor output is shifted a slightly different amount. This is called a phased array. A narrow band system, typical of radars or small microphone arrays, is one where the bandwidth is only a small fraction of the center frequency. With wide band systems this approximation no longer holds, which is typical in sonars.
In the receive beamformer the signal from each sensor may be amplified by a different “weight.” Different weighting patterns (e.g., Dolph-Chebyshev) can be used to achieve the desired sensitivity patterns. A main lobe is produced together with nulls and sidelobes. As well as controlling the main lobe width (the beam) and the sidelobe levels, the position of a null can be controlled. This is useful to ignore noise or jammers in one particular direction, while listening for events in other directions. A similar result can be obtained on transmission.
Beamforming techniques can be broadly divided into two categories:                a. conventional (fixed or switched beam) beamformers        b. adaptive beamformers or phased array                    i. desired signal maximization mode            ii. interference signal minimization or cancellation mode                        
Conventional beamformers use a fixed set of weightings and time-delays (or phasings) to combine the signals from the sensors in the array, primarily using only information about the location of the sensors in space and the wave directions of interest. In contrast, adaptive beamforming techniques generally combine this information with properties of the signals actually received by the array, typically to improve rejection of unwanted signals from other directions. This process may be carried out in either the time or the frequency domain.
As the name indicates, an adaptive beamformer is able to automatically adapt its response to different situations. Some criterion has to be set up to allow the adaption to proceed such as minimizing the total noise output. Because of the variation of noise with frequency, in wide band systems it may be desirable to carry out the process in the frequency domain.
Beamforming can be computationally intensive.
Beamforming can be used to try to extract sound sources in a room, such as multiple speakers in the cocktail party problem. This requires the locations of the speakers to be known in advance, for example by using the time of arrival from the sources to mics in the array, and inferring the locations from the distances.
A Primer on Digital Beamforming by Toby Haynes, Mar. 26, 1998 http://www.spectrumsignal.com/publications/beamform_primer.pdf describes beam forming technology.
According to U.S. Pat. No. 5,581,620, the disclosure of which is incorporated by reference herein, many communication systems, such as radar systems, sonar systems and microphone arrays, use beamforming to enhance the reception of signals. In contrast to conventional communication systems that do not discriminate between signals based on the position of the signal source, beamforming systems are characterized by the capability of enhancing the reception of signals generated from sources at specific locations relative to the system.
Generally, beamforming systems include an array of spatially distributed sensor elements, such as antennas, sonar phones or microphones, and a data processing system for combining signals detected by the array. The data processor combines the signals to enhance the reception of signals from sources located at select locations relative to the sensor elements. Essentially, the data processor “aims” the sensor array in the direction of the signal source. For example, a linear microphone array uses two or more microphones to pick up the voice of a talker. Because one microphone is closer to the talker than the other microphone, there is a slight time delay between the two microphones. The data processor adds a time delay to the nearest microphone to coordinate these two microphones. By compensating for this time delay, the beamforming system enhances the reception of signals from the direction of the talker, and essentially aims the microphones at the talker.
A beamforming apparatus may connect to an array of sensors, e.g. microphones that can detect signals generated from a signal source, such as the voice of a talker. The sensors can be spatially distributed in a linear, a two-dimensional array or a three-dimensional array, with a uniform or non-uniform spacing between sensors. A linear array is useful for an application where the sensor array is mounted on a wall or a podium talker is then free to move about a half-plane with an edge defined by the location of the array. Each sensor detects the voice audio signals of the talker and generates electrical response signals that represent these audio signals. An adaptive beamforming apparatus provides a signal processor that can dynamically determine the relative time delay between each of the audio signals detected by the sensors. Further, a signal processor may include a phase alignment element that uses the time delays to align the frequency components of the audio signals. The signal processor has a summation element that adds together the aligned audio signals to increase the quality of the desired audio source while simultaneously attenuating sources having different delays relative to the sensor array. Because the relative time delays for a signal relate to the position of the signal source relative to the sensor array, the beamforming apparatus provides, in one aspect, a system that “aims” the sensor array at the talker to enhance the reception of signals generated at the location of the talker and to diminish the energy of signals generated at locations different from that of the desired talker's location. The practical application of a linear array is limited to situations which are either in a half plane or where knowledge of the direction to the source in not critical. The addition of a third sensor that is not co-linear with the first two sensors is sufficient to define a planar direction, also known as azimuth. Three sensors do not provide sufficient information to determine elevation of a signal source. At least a fourth sensor, not co-planar with the first three sensors is required to obtain sufficient information to determine a location in a three dimensional space.
Although these systems work well if the position of the signal source is precisely known, the effectiveness of these systems drops off dramatically and computational resources required increases dramatically with slight errors in the estimated a priori information. For instance, in some systems with source-location schemes, it has been shown that the data processor must know the location of the source within a few centimeters to enhance the reception of signals. Therefore, these systems require precise knowledge of the position of the source, and precise knowledge of the position of the sensors. As a consequence, these systems require both that the sensor elements in the array have a known and static spatial distribution and that the signal source remains stationary relative to the sensor array. Furthermore, these beamforming systems require a first step for determining the talker position and a second step for aiming the sensor array based on the expected position of the talker.
A change in the position and orientation of the sensor can result in the aforementioned dramatic effects even if the talker is not moving due to the change in relative position and orientation due to movement of the arrays. Knowledge of any change in the location and orientation of the array can compensate for the increase in computational resources and decrease in effectiveness of the location determination and sound isolation. An accelerometer is a device that measures acceleration of an object rigidly inked to the accelerometer. The acceleration and timing can be used to determine a change in location and orientation of an object linked to the accelerometer.
U.S. Pat. No. 7,415,117 shows audio source location identification and isolation. Known systems rely on stationary microphone arrays. Known systems rely on stationary microphone arrays. In digital recording, audio signals are converted into a stream of discrete numbers, representing the magnitude of the audio air pressure or changes over time in air pressure. In this way, analog audio signals are converted into a stream of discrete numbers, representing the changes over time in air pressure. The discrete numbers are then recorded to digital media, such as DAT or addressable memory. To play back a digital recording, the numbers are retrieved and converted back into their original analog waveforms.
U.S. Pat. No. 7,492,907 B2 relates to multi-channel audio enhancement system for use in recording and playback and methods for providing same. It describes an audio enhancement system and method for use that receives a group of multi-channel audio signals and provides a simulated surround sound environment through playback of only two output signals. The group of audio signals, represent sounds existing in a 360 degree sound field, are combed to create a pair of signals which can accurately represent the 360 degree sound field when played through a pair of speakers. The multi-channel audio signals comprise a pair of front signals intended for playback from a forward sound stage and a pair of rear signals intended for playback from a rear sound stage. The front and rear signals are modified in pairs by separating an ambient component of each pair of signals from a direct component and processing at least some of the components with a head-related transfer function. Processing of the individual audio signal components is determined by an intended playback position of the corresponding original audio signals. The individual audio signal components are then selectively combined with the original audio signals to form two enhanced output signals for generating a surround sound experience upon playback
Ultrasounds are sound waves with frequencies higher than the upper audible limit of human hearing. Ultrasound is not different from ‘normal’ (audible) sound in its physical properties, only in that humans cannot hear it. This limit varies from person to person and is approximately 20 kilohertz (20,000 hertz) in healthy, young adults. Ultrasound devices operate with frequencies from 20 kHz up to several gigahertz.
Ultrasound is used in many different fields. Ultrasonic devices are used to detect objects and measure distances. Ultrasound imaging or sonography is often used in medicine. In the nondestructive testing of products and structures, ultrasound is used to detect invisible flaws. Industrially, ultrasound is used for cleaning, mixing, and to accelerate chemical processes. Animals such as bats and porpoises use ultrasound for locating prey and obstacles. Scientist are also studying ultrasound using graphene diaphragms as a method of communication. https://en.wikipedia.org/wiki/Ultrasound [11/24/2015]
Use of ultrasound to transmit data signals has been discussed. Jiang, W., “Sound of silence”: a secure indoor wireless ultrasonic communication system, School of Engineering—Electrical & Electronic Engineering, UCC, Snapshots of Doctoral Research at University College Cork 2014, http://publish.ucc.ie/boolean/pdf/2014/00/09-jiang-2014-00-en.pdf, retrieved Nov. 24, 2015. Sound is a mechanical vibration or pressure wave that can be transmitted through a medium such as air, water or solid materials. Unlike radio waves, sound waves are regulation free and do not interfere with wireless devices operating at radio frequencies. According to Jiang, there are also no known adverse medical effects of low-energy ultrasound exposure. On the other hand, ultrasound can be confined easily due to the way that it moves. Ultrasound travelling through air does not penetrate through walls or windows. Jiang proposes to use ultrasonic technology for secure and reliable wireless networks using digital transmissions by turning a transmitter on and off where the presence of an ultrasonic wave represents a digit ‘1’ and its absence represents a digit ‘0’. In this way Jiang proposes a series of ultrasound bursts travelling as pressure waves through the air. A receiving sensor may detect corresponding changes of sound pressure, and converts it into an electrical signal.
A voice frequency (VF) or voice band is one of the frequencies, within part of the audio range that is used for the transmission of speech. In telephony, the usable voice frequency band ranges from approximately 300 Hz to 3400 Hz. It is for this reason that the ultra-low frequency band of the electromagnetic spectrum between 300 and 3000 Hz is also referred to as voice frequency, being the electromagnetic energy that represents acoustic energy at baseband. The bandwidth allocated for a single voice-frequency transmission channel is usually 4 kHz, including guard bands, allowing a sampling rate of 8 kHz to be used as the basis of the pulse code modulation system used for the digital PSTN. Per the Nyquist-Shannon sampling theorem, the sampling frequency (8 kHz) must be at least twice the highest component of the voice frequency via appropriate filtering prior to sampling at discrete times (4 kHz) for effective reconstruction of the voice signal.
The voiced speech of a typical adult male will have a fundamental frequency from 85 to 180 Hz, and that of a typical adult female from 165 to 255 Hz. Thus, the fundamental frequency of most speech falls below the bottom of the “voice frequency” band as defined above. However, enough of the harmonic series will be present for the missing fundamental to create the impression of hearing the fundamental tone. Wikipedia, Voice Frequency, https://en.wikipedia.org/wiki/Voice_frequency, retrieved Nov. 24, 2015.
U.S. Pat. No. 3,806,919 entitled, “Light Organ,” is expressly incorporated by reference herein. U.S. Pat. No. 3,806,919 relates to a light organ and shows a system for energizing lights in response to sound intensity. Light organs may be responsive to a microphone or electrical signals corresponding to audio. U.S. Pat. No. 3,806,919 shows a detector amplifier stage that generates a signal representative of sound intensity detected by a microphone. The output of the amplifier stage controls the switching of a phase-controlled power switch connected across one of two lamp filaments connected in series. As the intensity of one lamp increases with sound intensity, the intensity of the other decreases. Automatic gain control circuitry adjusts the gain of the amplifier stages such that the lighting effect is substantially the same response for sound changes, and it is independent of ambient sound level. The lamps used are disclosed as having filaments which operate across an AC power source such as a full wave rectified 117-volt, 60Hertz source.
In various lighting applications, the use of light emitting diodes (LEDs) for illumination or decoration is now known. LEDs have long life, are energy efficient, are durable and operate over a wide temperature range. PixMob offers a wireless lighting technology that controls wearable LED devices intended to be worn by many individuals in a densely populated area such as a stadium or arena. By transforming the wearable objects into pixels, the crowd becomes a display. The light effects produced by the LED devices can be controlled to match a light show, pulsate in sync with the music, react to the body movement, etc. PixMob technology uses infrared or Bluetooth Low Energy (“BLE”) to control RGB LEDs that are embedded in different objects such as balls or wristbands. These wearable objects are given to an audience, transforming each individual into a pixel during the show. To light up each pixel (i.e. each LED), commands are sent from computers to transmitters that emit invisible light (infrared) or BLE. The signals are picked up by receivers in each object and goes to a microprocessor to control the LEDs. This enables the creation of animated video effects and transforms the audience into a display screen. Despite the low-resolution result due to a low number of pixels, quite detailed video effects can be achieved on a large canvas, using bright colors and bold movements. The control of an individual LED may be either based on an expected location of the LED or may be dependent on proximity to a known location.
Xylobands are another known wearable LED and control system for use, for example, in a concert venue. Xylobands are wristbands which contain light-emitting diodes and radio frequency receivers. The lights inside the wristband may be controlled by a software program, which sends signals to the wristband, instructing it to light up or blink, for example. They are available in green, blue, yellow, red, pink and white. The wristbands themselves may be constructed of a thick fabric with LEDs inside the fabric. A radio receiver is located within a plastic piece on the band, and it receives wireless signals from a controller, which is hosted on a laptop computer linked to a radio transmitter, which can remotely control the bands from up to 328 yards away. The operator of the laptop software may program all wristbands or only those of certain colors to flash on and off at specific intervals and specific moments. The wristbands are not intended to be lit outside of the concert venue. https://en.wikipedia.org/wiki/Xyloband.
U.S. 2014/0184386 A1 relates, in general, to an interactive lighting effect and is particularly, but not exclusively, applicable to electronic wristbands that can be selectively activated to energize light emitting devices integrated into each wristband to produce a coordinated display from individual wristbands worn by members of an audience at a show, such as a concert or a sporting event. In the exemplary context of an RF-based LED wristband with an integrated antenna. The wristbands are intended to be distributed at an event upon payment to an event organizer or pre-delivered. Typically, the wristband will include a controller coupled to a local power source, such as a battery. The controller is programmable through a suitable interface, which may include a physical connection or a passively accessible contact. In addition, each wristband contains at least one high-intensity LED device (or other controllable light-emitting device) operationally responsive to a control signal issued by a control station. The control station communicates with the wristbands using an RF transmitter and, if necessary, repeater stations that provide appropriate RF coverage within an arena or venue. Data bursts may be targeted using an activation code assigned to one or more of the wristbands. The wristbands may be assigned a zone address correspondingly the section of the venue that the user is expected to be in before it is deployed. Actuation of LEDs on the wristbands to support lighting effects is based on the assigned address and is not dependent on the actual location of the wristband in any way. The use of RF is preferred.
WO 2014/096861 A2 relates to a system for controlling light devices in a venue to create an image based on the position of the light devices. The position of a light device may be determined by GPS data or proximity using near field technology, RFID tags, or Bluetooth Low Energy devices such as i Beacons (RTE). Data indicative of the position of the pixel device is received at a server, a display attribute is calculated based on the position. This is particularly useful where the pixel devices are devices without a fixed position, such as mobile phones, PDAs and tablets, etc. for forming complex visual effects.